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chan_sip: Add precautionary p->owner checks.

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* Add precautionary p->owner checks in sip_hangup(), get_refer_info(),
get_also_info(), and interpret_t38_parameters().

* Simplify some tangled logic in get_refer_info(), get_also_info(), and
add_rpid().

* Removed some dead code in handle_request_invite().

(closes issue ASTERISK-23323)
Reported by: Walter Doekes
Patches:
      issueA23323-more_p_owner_checks-1.8.x.patch (license #5674) uploaded by wdoekes (modified)
      issueA23323-more_p_owner_checks-11.x.patch (license #5674) uploaded by wdoekes (modified)
      issueA23323-more_p_owner_checks-12.x.patch (license #5674) uploaded by wdoekes (modified)
      issueA23323-more_p_owner_checks-trunk.patch (license #5674) uploaded by wdoekes (modified)
........

Merged revisions 409207 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 409255 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@409256 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Richard Mudgett
2014-02-28 21:40:54 +00:00
parent 370b75e4e1
commit c38f3bd145
1 changed files with 42 additions and 40 deletions
Download Patch File Download Diff File Expand all files Collapse all files

66
channels/chan_sip.c
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@@ -7173,10 +7173,12 @@ static int sip_hangup(struct ast_channel *ast)
ast_debug(4, "SIP Transfer: Not hanging up right now... Rescheduling hangup for %s.\n", p->callid);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
ast_clear_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Really hang up next time */
if (p->owner) {
ast_channel_tech_pvt_set(p->owner, dialog_unref(ast_channel_tech_pvt(p->owner), "unref p->owner->tech_pvt"));
sip_pvt_lock(p);
sip_set_owner(p, NULL); /* Owner will be gone after we return, so take it away */
sip_pvt_unlock(p);
}
ast_module_unref(ast_module_info->self);
return 0;
}
@@ -7205,7 +7207,7 @@ static int sip_hangup(struct ast_channel *ast)
stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
append_history(p, needcancel ? "Cancel" : "Hangup", "Cause %s", p->owner ? ast_cause2str(p->hangupcause) : "Unknown");
append_history(p, needcancel ? "Cancel" : "Hangup", "Cause %s", ast_cause2str(p->hangupcause));
/* Disconnect */
disable_dsp_detect(p);
@@ -7735,7 +7737,9 @@ static int interpret_t38_parameters(struct sip_pvt *p, const struct ast_control_
AST_SCHED_DEL_UNREF(sched, p->t38id, dialog_unref(p, "when you delete the t38id sched, you should dec the refcount for the stored dialog ptr"));
parameters.max_ifp = ast_udptl_get_far_max_ifp(p->udptl);
parameters.request_response = AST_T38_REQUEST_NEGOTIATE;
if (p->owner) {
ast_queue_control_data(p->owner, AST_CONTROL_T38_PARAMETERS, &parameters, sizeof(parameters));
}
/* we need to return a positive value here, so that applications that
* send this request can determine conclusively whether it was accepted or not...
* older versions of chan_sip would just silently accept it and return zero.
@@ -12656,9 +12660,9 @@ static int add_rpid(struct sip_request *req, struct sip_pvt *p)
{
struct ast_str *tmp = ast_str_alloca(256);
char tmp2[256];
char *lid_num = NULL;
char *lid_name = NULL;
int lid_pres = AST_PRES_NUMBER_NOT_AVAILABLE;
char *lid_num;
char *lid_name;
int lid_pres;
const char *fromdomain;
const char *privacy = NULL;
const char *screen = NULL;
@@ -12669,22 +12673,20 @@ static int add_rpid(struct sip_request *req, struct sip_pvt *p)
return 0;
}
if (p->owner) {
connected_id = ast_channel_connected_effective_id(p->owner);
if (connected_id.number.valid) {
lid_num = connected_id.number.str;
if (!p->owner) {
return 0;
}
if (connected_id.name.valid) {
lid_name = connected_id.name.str;
connected_id = ast_channel_connected_effective_id(p->owner);
lid_num = S_COR(connected_id.number.valid, connected_id.number.str, NULL);
if (!lid_num) {
return 0;
}
lid_name = S_COR(connected_id.name.valid, connected_id.name.str, NULL);
if (!lid_name) {
lid_name = lid_num;
}
lid_pres = ast_party_id_presentation(&connected_id);
}
if (ast_strlen_zero(lid_num))
return 0;
if (ast_strlen_zero(lid_name))
lid_name = lid_num;
fromdomain = S_OR(p->fromdomain, ast_sockaddr_stringify_host_remote(&p->ourip));
lid_num = ast_uri_encode(lid_num, tmp2, sizeof(tmp2), ast_uri_sip_user);
@@ -18326,13 +18328,15 @@ static int get_refer_info(struct sip_pvt *transferer, struct sip_request *outgoi
}
/* Determine transfer context */
if (transferer->owner) /* Mimic behaviour in res_features.c */
transfer_context = pbx_builtin_getvar_helper(transferer->owner, "TRANSFER_CONTEXT");
if (transferer->owner) {
/* By default, use the context in the channel sending the REFER */
transfer_context = pbx_builtin_getvar_helper(transferer->owner, "TRANSFER_CONTEXT");
if (ast_strlen_zero(transfer_context)) {
transfer_context = S_OR(ast_channel_macrocontext(transferer->owner),
S_OR(transferer->context, sip_cfg.default_context));
transfer_context = ast_channel_macrocontext(transferer->owner);
}
}
if (ast_strlen_zero(transfer_context)) {
transfer_context = S_OR(transferer->context, sip_cfg.default_context);
}
ast_string_field_set(refer, refer_to_context, transfer_context);
@@ -18386,14 +18390,18 @@ static int get_also_info(struct sip_pvt *p, struct sip_request *oreq)
if (sip_debug_test_pvt(p))
ast_verbose("Looking for %s in %s\n", c, p->context);
if (p->owner) /* Mimic behaviour in res_features.c */
transfer_context = pbx_builtin_getvar_helper(p->owner, "TRANSFER_CONTEXT");
/* Determine transfer context */
if (p->owner) {
/* By default, use the context in the channel sending the REFER */
transfer_context = pbx_builtin_getvar_helper(p->owner, "TRANSFER_CONTEXT");
if (ast_strlen_zero(transfer_context)) {
transfer_context = S_OR(ast_channel_macrocontext(p->owner),
S_OR(p->context, sip_cfg.default_context));
transfer_context = ast_channel_macrocontext(p->owner);
}
}
if (ast_strlen_zero(transfer_context)) {
transfer_context = S_OR(p->context, sip_cfg.default_context);
}
if (ast_exists_extension(NULL, transfer_context, c, 1, NULL)) {
/* This is a blind transfer */
ast_debug(1, "SIP Bye-also transfer to Extension %s@%s \n", c, transfer_context);
@@ -25615,12 +25623,6 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, str
ast_debug(2, "No SDP in Invite, third party call control\n");
}
/* Queue NULL frame to prod ast_rtp_bridge if appropriate */
/* This seems redundant ... see !p-owner above */
if (p->owner)
ast_queue_frame(p->owner, &ast_null_frame);
/* Initialize the context if it hasn't been already */
if (ast_strlen_zero(p->context))
ast_string_field_set(p, context, sip_cfg.default_context);